Tag Archives: recording studio

Earlier this year I visited Steve Albini’s Chicago studio, Electrical Audio, with the goal of not only recording some drums with the man himself, but also scrutinising his mic techniques in order to learn more about how such an incredible drum sound is achieved. The results were as fantastic as you would expect, and I publicly documented my experiences via a video presentation, thanks in no small part to the assistance of an impressively bearded cameraman by the name of Kevin Clarke. You can see the resultant video here.

In order to record the session as flexibly as possible, and preserve all naked, ungrouped signals for my scrutiny later on, I knew that I had to split the signals coming from each microphone, with one batch going to Steve for recording to tape, and another identical but independent batch leading to my digital recording system. To make this work, Steve carried out an impressive feat of ad-hoc patching in order that we could split the signals after the desk preamp. This way I would still maintain whatever Neotek goodness was being imparted on each signal. The only differing variables between the two simultaneous recordings were the recording mediums themselves; Steve recorded to RTM900 2” tape, 16 track, 15 IPS. I recorded digitally at 48 KHz, 24 bit via a pair of chained MOTU 828 mk2 interfaces.

Now, I didn’t particularly give much thought to what interface I would use for my end of the recording, given that I consider all interfaces to be much of a muchness. They all do the same thing, and they all sound pretty much as transparent as each other. Quibbling about spec sheets aside, the fact of the matter is that the analogue-to-digital converters inside all interfaces across the price spectrum these days are perfectly capable of capturing and reproducing music transparently, and any talk about the correlation between “sound quality” and price tag is, in my opinion, grounded in a whole host of psychological biases which influence our perception of “quality” to an impressive degree, even more so when you’ve actually paid over the odds for what is, at best, an imperceptibly subtle improvement. “Yeah, this shit sounds fucking sweet. Now get an awesome photo of it for the website.”

The association of price with quality is a well-documented phenomenon, and companies like Apple and Neumann are masters of its manipulation. Of course you’d pay £65 for a MacBook charger! That’s the price you pay for quality (or a flimsy piece of shit that breaks after a year). And of course you would pay £250 for a U87 cradle! Why, you’d be an unprofessional fool not to (despite the fact that any old £10 cradle would do an identical job, they just lack the correct microphone attachment). Companies have been selling us overpriced shit for years, and the justification lands largely on the credos that a particular brand name has within their particular market.

Ah, but hang on… I’m not taking into account that, when it comes to audio devices, engineers the world over can really, really hear the difference! No, really! They really can! And that’s why they’ve got Prism converters in their studios! Because their ears are, like, super mega awesome. Better than your ears, and definitely better than mine! They sleep every night in an anechoic chamber in order to recalibrate their hearing for the day ahead, and the really top guys have bionic aural implants that allow them to hear all the way up to 40 KHz! They’re like dogs! They’re literally just like fucking dogs.

The fact that so many people claim to perceive an improvement in sound quality with respect to the price of their ADCs is, to say the least, unconvincing. There’s an interesting Caltech study that actually explores this phenomenon in more depth, which you can find on their website here. I’m going to lift the gist of the study from this website, which does a fine job of summarising it:

Researchers from CalTech and Stanford told subjects that they were drinking five different varieties of wine and informed them of the prices for each as they drank. But in reality, they only tasted three types, because two were offered twice: a $5 wine described as costing $5 and $45, and a $90 bottle presented as $90 and $10. (There was also a $35 wine with the accurate price given.) Not only did the subjects rate identical wines as tasting better when they were told they were pricier, but brain scans showed greater activity in a part of the brain known to be related to the experience of pleasure. In other words, the experiment may be evidence that we genuinely experience greater pleasure from an identical object when we think it costs more.

Admittedly not conclusive findings, it nonetheless points to a phenomenon that our deepest intuitions surely corroborate; when we think something costs more, or when we have a significant vested interest, we are biased towards defending its greatness.

So anyway, back to the Albini session. Based on what I considered to be an obvious truth about the perceived sound quality of over-priced ADCs, I was happy to facilitate my recording with the closest, most convenient solution to hand; a couple of MOTU 828s that I could easily chuck in a suitcase and get to Chicago without much hassle. However, in doing that, and knowing that these devices would appear in the video, I absolutely knew that some twat would appear from the woodwork at some point and feel the need to start the tedious “MOTU converters aren’t good enough” debate, and sure enough, one plucky commenter on my YouTube channel decided to do just that.

And so the criticism raged about how “MOTU converters sound cloaked and muddy” (whatever that means), and no engineer worth his salt would use anything less than Prism Orpheus converters at £ several-hundred per channel to capture anything like the kind of magic that Steve was laying down on his Studer A820, and he should know, because he’s worked in, like, loads of studios n’ shit, and, like, all his engineer mates agree… and this one time, right, he transferred an album using MOTU converters, and then had to re-record the whole thing because MOTU converters are, like, just so shitty sounding and all cloaked and muddy and stuff.

Anecdotal accounts of this kind are unimpressive for reasons too numerous to mention. The only way to claim real knowledge of this sort, I would argue, is to have performed some pretty rigorous blind trials, ensuring that only the single variable under scrutiny is the thing in the signal chain to be altered. Or indeed, if you feel confident enough to make the claim that something sounds “cloaked and muddy”, that you have subjected this to some close scrutiny such that you can describe in less ambiguous terms the perceived character of signal degradation, and under what conditions it manifests.

So, as much as I recline from the implication that I’m writing an entire blog post just to prove one person on YouTube wrong, I do think it presents an interesting opportunity to run a few tests and explore the topic further, not least so that my own perceptions may be less clouded by mere rhetoric of this kind.

Right then, let’s get to it. The first obvious comparison to run is between Steve’s tape recording and my simultaneous digital recording. This would essentially be a comparison between ATM900 2” tape and 48 KHz, 24 bit MOTU digital. Now, there are a few issues with this comparison which are worth laying bare at the outset, the most notable of which being that the multi-track transfer from tape to computer was itself made using the MOTU devices. This will obviously upset the angry YouTubers of this world, as the claim then becomes that the very act of running the tape recordings through MOTU converters has itself imparted intractable MOTU ugliness upon the signal, and therefore the comparison is of limited utility. And yes, I would agree that ultimate scientific rigour is sadly absent from such a comparison. However, that being said, if we are to assume that the 2” tape itself is doing something uniquely magical to the signal (warm, punchy, creamy, soupy… whatever bullshit adjective you want to throw in there), then we should expect to hear some difference between that recording and a purely digital recording. If the MOTU devices have done something nasty to the signal, then it should have incurred such nastiness identically on both the original digital recording and on the tape transfers, and as such we should still be able to identify which one maintains a semblance of the analogue magic. At the very least we should be able to tell them apart. Of course, tape is identifiable by its hiss, so in order to make this comparison fair, I’ve artificially added some hiss to the digital recording.

Below this paragraph there are two files; one tape recording (digitally transferred), and one simultaneous digital recording. Both are multitrack mix-downs of a solo drum performance with matched processing – the analogue tape utilising Steve’s outboard (as described in the aforementioned video), and the digital using comparable in-the-box plugins. You are invited to download them and compare them. See if you can spot the difference. Which one has the analogue “magic”? If you email me at james [at] jamesmakesmusic.com I will happily tell you which one is which, although be warned that I don’t consider this a bullet-proof perception test, given that you still have a 50% chance of guessing it correctly. This is simply a little comparison to get us started:

So let’s now move on to the real test, and that is a test of the claim that MOTU converters sound “cloaked and muddy”, and that I am clearly a fool to be using them. I’m going to take my cue here from the brilliant Ethan Winer, acoustician and life-long debunker of audiophile claptrap, who has himself conducted an identical test to the one I am about to perform, but with Focusrite and Soundblaster converters in his sites, rather than MOTU. You can find his tests on his website here. Essentially, the logic works like this:

The claim is that ADC X sounds crappy in some way (“cloaked and muddy” in this case). Therefore by performing a loopback recording using the ADC in question, over a number of generations this crappiness should become more and more pronounced. A loopback recording simply means playing an audio file out of a stereo output of the device, and physically patching that output back into two inputs and recording it in whatever recording software you use. The resultant recording is then used as the source for the next loopback recording. And that one for the next. And that one for the next. And so on. After several generations the claim should be very easy to verify, as the signal degradation, or “cloaked muddiness” should be cumulatively imparted on the signal, such that the discrepancy between, say generation 10 and the original file is absolutely obvious. Certainly, if the ADCs under scrutiny are as crappy as has been claimed, then even after a single generation we should hear an obvious difference between the result and the original file. However, if it transpires that it is a struggle to hear any difference, and actually in a blind trial it is not even clear which is the original file and which are the subsequent generations, then it’s pretty safe to say that people like our YouTube friend may well just be subject to the aforementioned psychoacoustical biases, and therefore, simply talking shit.

Below this paragraph you will find several batches of files. I used four different pieces of music as my test subjects, ranging from grunge, through trip hop, to classical and jazz. For each genre I have posted four files; the original, generation 1, generation 5 and generation 10. I have aligned all recordings and gain matched as closely as possible. All recordings are carried out at 44.1 KHz, 16 bit, in order that any degradation manifests as obviously as possible. The goal for anyone who wants to partake in the challenge is to correctly identify each file. Once again, you can obtain the correct answers by sending me an email to james [at] jamesmakesmusic.com. If you are confident about MOTU converters being unsuitable for use because of their defects in “sound quality”, then you should have no trouble correctly identifying each file. And to RasTatum – the man who made the claim about MOTU converters sounding “cloaked and muddy” in the first place (but who has since rather curiously deleted all his comments) – I eagerly await your contribution to this experiment.

Finally, the last brief experiment I wished to perform was a test of the self-noise of MOTU converters, because whilst the sound quality may not be as affected as we thought, then that still leaves room for the claim that the devices themselves are noisy. So I performed another loopback test using a file of total silence as the source. I won’t bother actually posting the resultant audio files here, but I will tell you that after 20 generations I was seeing a noise floor of -72.2 dB, as you can see from the image below. I hope you’d agree that that demonstration renders concerns of this type absolutely negligible.

So there you have it. Turns out you don’t actually need to spend thousands of moneys on ADCs just because someone tells you to.

For the second time in my life I have realised that reaching out to a mastering studio to put the “finishing touches” on my music is completely pointless.

Allow me to explain…

Mastering recorded audio became its own discipline after the Second World War, when a “dubbing engineer”, secondary to the recording/mix engineer, was tasked with transferring the recorded audio from tape to a master disc, which served as the template from which all following vinyl discs would be pressed. This was a purely technical procedure, whereby the dubbing engineer’s job was to ensure that the final recording, which had been signed off by those creatively involved with the production of the music, was faithfully duplicated onto its designated medium.

Early vinyl records tended to be dogged by various inefficiencies in the tape-to-disc transfer process, not least that the dynamic range of the recorded material could be too large, resulting in the cutting of unplayable waveforms where the needle would actually pop out of the grooves, or even burning out the disc cutting head. The use of compressors and limiters in the mastering process became widespread in the 1960s, to cap the dynamic range at a particular threshold and thus ensure that such problems could be avoided. However, because this process was automated, often the dynamics processing employed was not sympathetic to the fidelity of the original material, and so over-compression would sometimes squeeze the life out of it, making everything sound consistently loud in a way that dishonoured the integrity of the original tape master. Some records ended up sounding particularly nasty due to this pitfall at the mastering stage.

And so, the solution to this problem?

Enter the mastering engineer.

By the 1970s, dedicated mastering studios had been established, staffed by sound engineers using high-end equipment. These “mastering engineers” were incredibly adept at finalising tape masters in an artistically satisfactory way, establishing mastering as a new artistic discipline that could actually make the final result sound “better” than the original recording.

Throughout the 80s and 90s, music production was revolutionised by digital technology, and CDs became the darling format of the music industry. To this end, the significance of mastering for vinyl became less prominent, as the problems incurred by analogue playback were no longer an issue in the digital domain. Mastering engineers, however, did not disappear, and instead their role migrated into audio specialists who serve as the last step in the production process – the guy or gal who collates all the final mixes for a particular release, and applies their technical wizardry to ensure that program volumes and tonal balancing are consistent throughout the entirety of the album. This is arguably of particular importance given the infinitely flexible DIY audio production world in which we now live, where track one may have been recorded and mixed in your bedroom, and track ten is a live recording from that gig you played last year – a far cry from the rigidly calibrated standards of professional audio recording of the 60s and 70s – the mastering engineer can be an invaluable specialist who coalesces all of these final mixes, “topping and tailing” each song to run seamlessly from one to the other, and thereby creating a pleasingly consistent album.

So, what’s my beef with mastering then? Why the need for such cynicism over a specialist process that seems so necessary?

Well, as we have seen, the discipline of mastering has migrated away from being a technical necessity, and has reinvented itself as an artistic process that seeks to “correct” and “improve” audio recordings. It seems to me that underpinning this is an assumption that all recordings require “correction” and “improvement”, such that it has now become an almost unquestioned assumption that recordings must undergo such processes before they are properly finished, regardless of the fact that 99% of all recordings these days end up uploaded onto Soundcloud or YouTube, and as such have absolutely no technical requirement for any fiddling at the final stage. I have had this demonstrated to me twice in my life, and both times I reached the conclusion that mastering is really only necessary if identifiable problems are present with the final mixes. In short, if your final mixes sound great to you, and you are satisfied that they translate well across systems, then you really have to ask yourself what the point of having it mastered actually is.

Case in point, I recently finished working on two songs of my own, and rather than do my normal thing of using some light compression, adding a little sweetening EQ and then normalising the result, I decided that it is high time I found myself a decent, trustworthy mastering engineer to whom I could reliably outsource any material recorded at my studio – for both myself and my clients – to put the “finishing touches” on the mixes. The icing on the cake. The cherry on top. The sachet in the pot noodle. The mayonnaise on your kebab. Whatever your favourite culinary analogy, that’s what I thought. And so I touched base with several mastering facilities, both home and abroad, each of whom did a test master for me of one of my songs.

The result?

In each instance I found their work to be a terrible detriment to my original mix; crushed with compression in a way that seemed to me to be horribly distasteful, and accompanied by notes claiming things like “I tried to make it a tad warmer and kill some spikiness in the guitar”. This seemed to me to be slightly presumptuous – perhaps I like the spikiness in the guitar (I do). But of course, how was he to know otherwise? He is not familiar with my style, my artistic preferences, or what I consider important about my mixes, and so he was just trying to rectify the problems in the mix, as he perceived them. Attempts to articulate my preferences via email just leads to a cumbersome back and forth whereby words prove to be an inefficient medium in which to convey the subjective pleasure of ambiguous terms such as “guitar spikiness”, let alone any other of the myriad things that I neglected to mention. I actually work hard to capture a wide, natural ambience in my music, especially in the drums, and I feel that, in this current age of “loudness war” style over-compression, excessive limiting of transients in order to push up the aggregate volume of the music actually works against this kind of production style, and forces a kind of “breathlessness” in the music, where everything becomes squashed into a mulch of muddy sounding loudness.

Let’s take a closer look…

The image above depicts a stereo waveform representation of my original mix (red), followed by two subsequent masters from two different studios. In both cases we see that the audio peaks have been truncated in order that the aggregate level can be further maximised. The blue-backed waveform represents an attempt by the first mastering engineer – this waveform is a real sausage! Obviously hugely compressed (oddly more so on the right hand side than the left), which manifests as very noticeable “gain pumping” (sharp volume rises and falls) when listening. Detailing this concern to the second mastering studio, they returned their master, which is the yellow-backed waveform above. Noting that I was not a fan of excessive compression, they opted to still squash the mix, but just not as much. The result was a slightly less severe but still noticeable and ugly compression.

It actually seems to have become second nature to mastering engineers to simply make everything as loud as possible, because, hey, louder = better, right? We can see this trend towards excessive loudness by comparing two more waveforms, this time from Nirvana’s song “Smells Like Teen Spirit”, recorded in 1991. The image below depicts the original 1991 master (blue), and the 20th anniversary remastered “Special Edition” from 2011 (green):

It is interesting to note that the blue-backed waveform clearly shows Nirvana’s signature loud-quiet-loud song structure represented as an actual change in peak volume between the verses and the choruses. Cut to 2011 and this natural dynamic has been crushed in order raise the aggregate level of the song, arguably sacrificing one of the very trademarks that made Nirvana such a dynamically versatile and intense band in the first place. So no, louder is not always better.

But here’s another reason to be wary of excessive compression. Look what happens when we truncate peak waveforms in this way:

The above image shows a close-up of my original mix (red) side by side with the first master (blue). What we see is that, by truncating audio transients we are actually sacrificing audio content that would otherwise have been present. The detail displayed in the red wave has been totally lopped off and replaced with something resembling a large square wave. Square waves actually introduce odd-ordered harmonics into the signal, which manifests to our ears as rather ugly distortion.

So it seems to me that we are somewhere close to the old days of ramming final mixes through limiters at the mastering stage simply as a matter of course rather than because the music actually warrants it. Indeed, when I suggested to subsequent mastering engineers that I don’t wish to overdo the compression, they still felt inclined to push it somewhat, rather than to err on the side of subtlety. It’s curious why this has become the norm, and of course the much discussed “Loudness War” of the 2000s has impacted significantly upon the industry, such that it seems as though a mastering engineer doesn’t feel he is creating value for money unless he is seen to be mastering for “competition volume”, or else tampering with the mix to some significant and obviously noticeable degree. But for me, this is not actually the job of a mastering engineer. It seems to me that a principled mastering engineer should not be afraid to listen to a mix and decide that nothing needed to be done to it. And to that end, their job is done, and they are still every bit as entitled to be paid as if they had actually decided that there were real tonal balance problems that needed to be rectified. The mastering engineer is your last line of defence against actual technical problems, not a dude who can make your mixes sound “shit hot”. Working under that preconception actually encourages sloppy mixing, because it’s okay – the mastering guy will fix it!

So, where does this leave me?

Well, just to be clear – I am not a mastering engineer, and I do not claim that I can adequately do the complicated job of fixing the technical problems of someone else’s mixes. This task is for dedicated mastering engineers who are good at what they do and conduct themselves in a principled and agreeable manner. But I would urge you, if you’re happy with your mixes and you love the way they sound, please ask yourself – what exactly is the problem that you’re trying to solve? Personally I can only conclude the same point that I reached several years ago when I went through a similar experience: I seem to be trying hard to locate a mastering engineer to whom I can pay money in order to fix unidentifiable problems. All they seem to do – inevitably – is fail to align with my artistic vision and return results that I actually think make my mixes sound worse, not better. And so, being that I do not wish to employ someone to make further creative decisions on mixes that I am already satisfied with, it seems to me that I should take my cue from my previous decision on this matter, and that is that the person best placed to put any “finishing touches” on my music is me.

Hopefully, in a few years from now, when I have again forgotten why I don’t use mastering engineers and I find myself once again looking for that special someone who can put the awesome “finishing touches” on my music, this blog post will serve as a reminder of just how pointless that pursuit is.

The topic of audio recording is vast and open-ended, and discussion about associated equipment in particular often gives rise to much heated debate with respect to perceived differences in the sonic performance between devices. It is not uncommon for hostile discussions to be waged pitting the minutia characteristics of this device against that, with all parties using increasingly elaborate language to define their subjective auditory experience, yet in the process obfuscating any real scientific analysis in favour of regurgitating “buzz” words that, when examined, actually fail to reveal anything helpful about the nature of the device in question. “Warmth”, “openness”, “air”, “punch”, “creaminess”, “sheen”, “silkiness”, “purpleness”, “dogturdidness”; fluffy terminology of this nature can often be observed in industry magazines (as some notable culprits are particularly guilty of), where vast word salads are served up in an attempt to suitably bewilder the reader into believing some imposed perception about a given piece of equipment. Whether it is an industry effort to create brand association with generic “good sounding” Barnum statements, or simply sloppy journalism in which authoritarianism comes from using words that everyone is too confused to question, the amount of bullshit I witness people talking on a regular basis goes to show how successful this method is.

I find language of that nature problematic for several reasons, not least because it denies us, as students of audio recording practices, access to scientific truths with regards to our field, where discussion of imparted harmonic content via signal distortion is much more helpful than fogging the issue under a linguistic cloud of subjective terminology and thereby propagating marketing myths about the necessity of over-priced equipment. It is no doubt a valuable weapon across all levels of the audio equipment industry, each brand justifying the apparent necessity of its newest model by using words that no one really understands. It’s interesting how readily we accept this lack of clarity in the discussion of audio, and how encourageable everyone seems to be to jump on the bullshit bandwagon. Note how we don’t accept this terminology in discussion of equipment where the scientific validity of their specifications really matters – I’m sure no FMRI scanner was sold on the basis of the “punch” of the scan or the “warmth” of the images produced. We can more readily accept fuzzy jargon in that context as obviously ridiculous and unhelpful.

“Brilliance”, anyone?

One of my audio recording heroes is Ethan Winer – musician, acoustician, and owner of the acoustic treatment company RealTraps – Ethan is somewhat notorious for his efforts to debunk tenacious myths prevalent among recording enthusiasts, whilst grounding his discussions in empirical scientific analyses, thereby abstaining from and often criticising the use of ambiguous subjective terms. I highly recommend his book “The Audio Expert” in which he talks about this very topic:

“Some of the worst examples of nonsensical audio terms I’ve seen arose from a discussion in a hi-fi audio forum. A fellow claimed that digital audio misses capturing certain aspects of music compared to analog tape and LP records. So I asked him to state some specific properties of sound that digital audio is unable to record. Among his list were tonal texture, transparency in the midrange, bloom and openness, substance, and the organic signature of instruments. I explained that these are not legitimate audio properties, but he remained convinced of his beliefs anyway. Perhaps my next book will be titled Scientists Are from Mars, Audiophiles Are from Venus.”

With this in mind then, allow me to demonstrate the principle of audio bullshit in action. As I came to undertake an investigation into the sonic differences between several different microphone preamps (post on that soon), I encountered a 2007 article from Sound On Sound in review of the Neve Portico 5012 Dual Microphone Preamp. As my curiosity led me to probe how such a device can justify a £1,400 price tag, one sentence in particular proved to be such an excellent demonstration of the ambiguity of industry terminology that I was inspired to finally write this blog post, hailing my discovery as a gold standard of audio bullshittery. Let’s have a look:

“The 5012 […] has a full bodied, solid sound that gives that slightly larger-than-life character that is the trademark of a really top-class preamp. It sounds clean and detailed in normal use, without that edgy crispness that can detract in some designs…

When the Silk mode is switched in, the sound becomes a little smoother, rounder, and sweeter still in the upper mids. The high end gains a little more air, and the bottom end becomes a tad richer and thicker.”

Terms like “larger-than-life” and “edgy crispness” are rampant when describing microphone preamps, analogue-to-digital converters and other studio essentials, yet they say nothing useful whatsoever about the actual, verifiable sonic characteristics of the device, instead simply propagating the usage of these vague terms and using them as flimsy justification for impressionable enthusiasts to feel anxious about the “below-par” consumer grade equipment they are currently using, and therefore encouraging them to unnecessarily part with not insignificant sums of money, thereby continuing the trend. That’s not to say of course that there is no value in “high-end” gear such as this, however I would prefer that its usage could be justified in more certain terms than these floppy, nothing words that we all have to keep grappling with. In my experience it’s always worth pushing for clarification via language that is arrived at through scientific consensus so that we can all be on the same page in terms of our expectations. This is the best prophylactic available against the tech-heads who claim authority by asserting that their £X,000 device sounds “sweet”. Chances are, they’re talking bullshit.

Often in recording scenarios it is necessary to implement a stereo miking technique. Usually this is employed to capture room ambience at a distance from the originating sound source, by which I mean the reverberant field of an acoustic environment – anywhere where the late reflections are of greater intensity than the direct sound. Whether it’s for drum kit ambience, concert halls or choirs, ambient stereo miking provides a way of adding depth, width and general realism to the recording that is not possible through close-miking alone.

However, there are numerous stereo mic techniques and it struck me recently that I had never undertaken a direct comparison of them. This realisation in fact struck me with such vigour that I felt moved to instantly rectify the situation, spontaneously leaping up from my seat, screaming “STEREO MIC COMPARISON!!”, and bolting, arms flailing and screeching like a girl, towards the door. The other cinema-goers were somewhat bemused.

And with that I decided at once to trial four different stereo mic techniques over a few different scenarios. These are techniques that any good engineer should be aware of, but perhaps not all have actually directly compared. Well, in the name of science I hereby rise to the challenge.

Yes, that’s right… science.

So the four techniques on the menu today are the following:

#1: XY

#2: ORTF

#3: Blumlein

#4: Mid/Side

I won’t go into detail about the configuration of these techniques here, largely because it’s late and I can’t be bothered, but if you’d like to know more about their implementation, please follow this link.

The purpose of my trials would be to answer the following questions:

Which technique captures a more effective and balanced stereo image?

How well does each technique collapse to mono?

How rich is the tonal balance?

Which one do I like best?

I chose to make these comparisons under 3 different scenarios: a large, reverberant concert hall, the drum recording environment in my studio, and with a moving sound source in a small room, which in this case was me wandering around and talking. The tests employed two sets of microphones – two AKG C414s for Blumlein and Mid/Side, and two AKG C451s for XY and ORTF. This selection was imposed due to equipment restrictions, otherwise identical microphones would have been used for all applications, thereby eliminating the variable of the sonic performance of the different mics. Nevertheless the comparisons should allow us to draw some reasonably solid conclusions.

Listed below are the recordings. Click each one to listen for yourself and see if you agree with my analysis:

So, based upon these recordings, along with a whole load of other tests I carried out which are not listed above, here are my answers to the aforementioned questions:

Q: Which technique captures a more effective and balanced stereo image?

A: Mid/Side.

The weakest stereo image seemed, across the board, to be XY. It has a strong centre but very little width. This is unsurprising, since the capsules are so close together that it seems illogical to expect anything more. This is as I had always suspected, and why I never really felt tempted by this technique. The lack of movement in the voice recording is particularly noteworthy. My next preference is ORTF due to its much wider stereo image and strong centre point, followed jointly by Blumlein and M/S, both of which clearly exhibit a wide, detailed image. If I had to pick a winner, I’d go with M/S. The movement may not be quite as authentic as Blumlein, perhaps due to the trickery involved in the M/S configuration vs. the fairly organic method of Blumlein, however for the capture of room ambience for a static source, M/S just seems to have a special kind of something about it – a width and depth that to my ears is incredibly realistic.

Q: How well does each technique collapse to mono?

A: ORTF & Blumlein win.

The mono drum recordings reveal that no technique has any particular issue or phase weirdness occurring when collapsed to mono, however in terms of preserving the fidelity of the ambient field that we are attempting to capture, Blumlein and ORTF seem to have it over M/S and XY. With M/S this is due to the cancellation of the side mic so that we are left with only one microphone pointing at the source, and XY had the weakest stereo width anyway, so this result is unsurprising.

How rich is the tonal balance?

A: Mid/Side wins.

We have to be a little careful here when we start using ambiguous terms like “richness”, “warmth”, “creaminess”, “silkiness”, “moistness”, “purpleness”, etc, etc, because these aren’t exactly scientific words. However, what I intend it to mean in this instance is how well expressed are the bass, middle and treble parts of the frequency spectrum, subjectively speaking. In my view, M/S clearly trumps all others in terms of its pleasing bottom end yet detailed high frequency reproduction. This was deduced by looping small parts of the drum and concert recordings and directly comparing each technique. ORTF is also very good in this regard, followed by Blumlein and finally XY.

Which one do I like best?

A: Mid/Side!

Yep, it would appear that M/S is awesome. Science says so. Well, to my ears at least. My science ears. It adds something magical to the recording and is extremely pleasing to experience, especially on drum recordings when combined with the close miked signals. Here is a demonstration of that:

Blumlein and ORTF are still excellent techniques though, offering a nice, solid centre and plenty of detailed width, which is certainly bad news for the XY technique, which has since been strapped to a rocket and jettisoned into the centre of the sun.

Recording drums in a small room is a problem that any engineer not blessed with an infinite budget must deal with at some point. Among the difficulties inherent in this scenario is the problem of comb filtering in the audio signal due to the microphone’s proximity to a boundary, i.e. the ceiling or a nearby wall. For example, if a singer sang into an omni-directional microphone placed 1 metre from a reflective wall or surface, the sound of their voice would hit the mic but also carry on past it, hit the wall, rebounding back and re-entering the mic about 6 milliseconds after the direct signal.

This is exactly the right amount of time for the frequency components around 85-86Hz to come back close to 180° out of phase with the direct signal. There will not be total cancellation, since the rebounded signal will be weaker and because the sonic characteristics of the singer’s voice are constantly changing, but the effect may still be significant.

Rounding down to 85Hz, at 170Hz the reflection will come back in phase and reinforce the 170Hz components within the direct signal. At 255Hz it will be out of phase again, and at 425Hz and 595Hz, and at intervals of 170Hz all the way up the frequency spectrum. This is known as “comb filtering”, due to the regular series of peaks and notches across the spectrum. It sounds phasey and generally undesirable.

This effect is demonstrated in this video, where a drum overhead microphone is moved towards a nearby boundary and back again. The comb filtering artefacts are clearly audible in the recorded signal. The first microphone – a Royer R121 ribbon mic – clearly suffers from this effect with great prominence given it’s bi-directional polar pattern, and thus greater susceptibility to rear reflections. The second mic – an Audio Technica ATM450 – reveals itself to be less harshly affected due to its cardioid polar pattern. This then demonstrates the importance of microphone selection with regard to its placement within a recording environment, as well as the importance of placing the mic as far from boundaries as possible, or, when this is not feasible, treating nearby surfaces with good quality acoustic absorption in order to eliminate as many reflections as possible. A combination of absorption and diffusion is most effective.

Many thanks to my beautiful assistant, Bebe Bentley, for helping me with these tests. Check out her excellent work in film and moving image on her Vimeo page.

Have you ever wondered how it is possible for the human brain to so accurately detect the location of a perceived sound? We only have two ears, yet somehow we are able to discern the differences between sounds originating from any direction within our 3-dimensional environment – in front, behind, above, below, left or right. How is this possible? And can we therefore simulate this effect in order to artificially reproduce the experience of perceived 3-dimensional sounds, as opposed to the normal left/right experience we are accustomed to in traditional stereophonic speaker set-ups, without simply adding extra speakers?

The answer is yes we can. Directional perception of sound occurs by our brain’s ability to decode the subtle differences in information received by our in-built stereo receivers – our left and right ears. Binaural recording is a recording technique that uses two microphones to mimic the human auditory system, utilising the exact same conditions that create the phenomenon of binaural localisation in humans. And so, with the acquisition of a pair of binaural microphones, a portable Tascam field recorder and a dummy head named John, film maker Bebe Bentley and I spent one evening carrying out some binaural recording tests at the University of Sussex. Here are the results (please note that headphones must be worn in order to perceive the effect):

In the directional perception of sound there are two phenomena at work: Binaural and monaural localisation:

Binaural Localisation

Binaural Localisation refers to the discrepancies in the characteristics of a sound wave arriving at the closest ear, and then the farthest. Your brain is sensitive to the discreet time difference between a sound hitting the nearest ear and the farthest ear – referred to as the Inter-aural Time Difference (ITD) – as well as the slight change in volume between the two ears – the Inter-aural Intensity Difference (IID). If sound originates to your left, your head acts as a barrier or filter and reduces the level of sound heard in the right ear.

Monaural Localisation

Monaural localisation mostly depends on the filtering effects of physical structures. In the human auditory system, these external filters include the head, shoulders, torso, and outer ear or “pinna”, and can be summarized as the head-related transfer function. Sounds are frequency filtered specifically depending on the angle at which they strike the various external filters.

Binaural recording of the kind Bebe and I carried out works by the use of two omni-directional microphones fitted to a dummy head, thereby simulating as realistically as possible the actual physical location of the human ears, combined with the filtering incurred by the human head. The same effect would be achieved by placing the microphones in your own ears, which would make for an interesting audio experience were you to then simply walk around an urban environment or visit a concert. In these instances it would be possible to accurately record exactly what you heard in these situations, complete with directional perception of the ambient noise, in order to later recreate that exact sensation through a pair of headphones. This, however, is perhaps a test for another day. Here we simply affixed the microphones into John’s ears and proceeded to move objects around and make various noises such that the illusion of directional perception is created.

It is however important, for the effect to be fully realised, that headphones are worn. This is because, on replay, the left ear must receive only the signal recorded by the left microphone, and the right ear only the signal from the right microphone. Playback through speakers destroys this effect by obscuring the stereo field emitted by the left and right speakers.

What strikes me as odd about the experience of listening to this recording is the realism it invokes. When hearing Bebe and I running around the room it is as if ghost figures are appearing in front of you. With your eyes closed you can almost “see” the people. This demonstrates just how unaware we are of the subtleties of our sensory information in building our picture of the world. The next time someone supposes some supernatural bullshit to describe how they “felt a presence in the room”, remind them how easily our senses can be fooled.

So there we are. Artificial directional perception by binaural recording. Now, if only I could find a practical application…

When recording a drum kit one of the most perennial problems encountered is high-hat spill on the snare microphone. Some engineers claim to have made peace with this issue by utilising the signal as simply “part of the drum sound”. This doesn’t do it for me since, among other problems, it ruins my stereo image of the kit, placing the hats immovably in the centre. Others aim their microphones such that the null in the cardioid pattern (i.e. the rear of the mic) is directed at the hats. Others even suggest using a figure-8 mic such as a ribbon, which has deeper nulls in its off-axis response, placed so that the side of the capsule looks at the hats.

None of these solutions provide suitable buoyancy to float my little boat. For a start, dynamic mics – especially the SM57 – do not, in my opinion, sufficiently capture the snap and sizzle of a snare drum, and besides, positioning one so that its rear is pointing towards the hats without disturbing the drummer is a tactical nightmare. Ribbon mics are scarcely much better, since there is no one location where the rear of the microphone is not detecting an unworkable amount of the tom behind it. And I don’t even want to think about the consequences of the inevitable battering it is going to take from the drummer. In any case, microphone positioning of this nature when in such close proximity to other undesirable sound sources is purely a hypothetical exercise. In the real world the results achieved by nit-picking in this manner are more or less negligible. The harsh spill from a close set of loud high-hats is simply not going to be significantly reduced by inching a microphone on its axis one way or another.

When I record snare drums I generally like to use the very tiny Shure Beta 98 microphone. It sounds absolutely excellent, gives great top end crack, has very fast transient response, and is so physically small that it can be positioned anywhere around the drum without getting in the drummer’s way (it also has great mounting hardware so as to clamp rigidly onto the side of the drum, thus eliminating the requirement of yet another mic stand). Then when I mix the snare I like to take a good, transparent EQ and make it extremely bright. That’s how to achieve a good crack that pierces like a razor blade though the mix. However, in order for this to work the snare must be as isolated as possible from the rest of the kit, and the high-hat above all must be eliminated as much as possible from the signal, or at the very least its high frequencies significantly reduced.

So. We have a conundrum on our hands. If we can’t budge on mic choice and we can’t solve the problem through placement, the only other alternative is baffling. With this, I set to work.

Now, I have read several times on forums and in textbooks such as Bobby Owsinsky’s “The Recording Engineer’s Handbook” that a good method of baffling ambient sound from a drum mic is to cut a hole in a polystyrene cup, poke your microphone through the middle and then tape the contraption together. Dubious, I gave it a try, suspecting that polystyrene does not present a suitably absorbent or reflective material to deflect close proximity, high intensity sound. As it transpires, I was right. Not only this, but I couldn’t imagine actually putting this into practice in a recording session without feeling like the dickiest of amateur dicks: “We’re all miked up lads… now, get me a paper cup and some gaffer tape!”. However, somewhat inspired by this idea I thought that perhaps I could build a contraption out of a more rigid material, take some steps to furnish it with some proper isolation material and then affix it retractably to the microphone, thus making for a more professional, more effective baffle and thereby solving our problem.

The idea? Tennis balls! One tennis ball, in fact. Cut in half, a hole cut in the middle, the outside covered in tin foil and the inside stuffed with acoustic foam. As I sat in one sunny Saturday, craft materials sprawled everywhere and glitter all over my face, my train of thought pulled in for a long stay at Genius Junction. This, I knew, was the solution to all my high-hat woes. I was indeed a genius. The result looked like this:

I thought it looked pretty smart. But did it work? Well, let me tell you…

No. It was shit. Not only was it absolutely ineffective, it also turned the source, i.e. the snare, into a tonally retarded shadow of its former self. And this makes perfect sense too – if you place a microphone within the confines of a cavity, then the acoustical properties of that immediate boundary are going to wreak havoc on the direct source you are trying to capture. The resonant frequency of that cavity combined with the filtering artefacts incurred by the boundary (the boundary effect) are going to dick with your source sound in a totally undesirable way. To see for yourself, just cut a hole in the bottom of a paper cup and put it up to your ear while listening to some music. Sounds awful, doesn’t it? If more proof were needed, here are the results of my tests:

So I think we can safely say that forming any kind of cavity immediately around a microphone is definitely not a good idea. This means that we have to find some other non-intrusive way of baffling the high-hats. Since the tennis ball idea not only sounded bad but also did very little to reduce the harsh frequencies of the hats, it seemed to me that we needed to think bigger to think better. I know from experience that an extremely good source of acoustic insulation is Rockwool, due to its high absorption coefficient, especially in the high frequencies – exactly where the harshness of the hats resides. So if we could somehow fashion a non-intrusive baffle out of four inches of Rockwool, then maybe we would be on to something. I immediately got to work on some leftover sound insulation with a Stanley knife. After many hours chopping, changing and inhaling an ever increasing quantity of microfibres, I discovered a solution that created no cavity around the microphone and significantly reduced the harsh top end of the hats in the snare mic. That solution was to raise the hats such that a four inch thick slab of high-hat shaped Rockwool could be installed beneath them, with the snare mic tucked underneath. It wasn’t pretty but it worked a treat:

For those of you with anxieties about raising high-hats, I should point out that this approach really is the first port of call when attempting to reduce high-hat spill. The further away you can move a source from the microphone, the less intrusive it will be. With the hats this carries the added bonus that it moves the drummer’s point of contact to the less clangy side of the hats, as opposed to the harsher top.

Finally we’re getting somewhere. For good measure, and simply because it seemed like it was something I should do, I added a chunk of acoustic foam underneath the Rockwool, just to see if I could knock off that spill a little more:

The results were excellent. The high-hat spill was becoming reduced to a much more manageable level:

The only remaining problems now were a) how to make this monstrosity more aesthetically pleasing, and b) how to not disrupt the drummer by its presence. Both of these concerns were addressed by cutting the Rockwool down to exactly the size of the high-hat (generally 14″) and taking one extraordinarily tedious afternoon to assemble a small pair of trousers in which to house it all:

The Rockwool was inserted into the black cotton trousers, with the foam glued to the underside. By clipping this to the stand immediately beneath the hats, the microphone can can be tucked discreetly underneath, also then protecting the mic from an accidental battering from the drummer.

And there it is! This is how to eliminate high-hat spill without ruining your snare sound. And it just goes to show – don’t just believe what the textbooks tell you. Try it yourself, and if it doesn’t work, get creative.

While working as a freelance engineer in a prominent Brighton studio I saw that here lay an excellent opportunity to properly exploit a decent live room and a large selection of microphones (a combination of ribbon mics with my own extensive set of small diaphragm condensers) in order to once and for all produce the kind of drum sound that provokes wet dreams in any fan of Steve Albini, and probably be the only studio in Brighton with the foresight to do so. With that notion, and with the promise of a kick drum sound capable of producing an internal haemorrhage elusively wafting through the air, I set to work in order to discover exactly how it is done, and exactly which mics and techniques are appropriate to induce such biological phenomena in a laborious weekend of tedious excitement. Aided by fellow engineer Chris Blakey, and musician and professional cable winder Genti Aliaj, these were our findings.

The Approach

All tests were conducted through a Neve 5316 using the console preamps, recorded to Pro Tools 9 HD at 96 KHz, 24 bit, and monitored through NS-10Ms and KRK VXT6s.

In order that the best drum sound could be obtained, the approach was logical and methodical:

Get a decent sounding drum kit.

Tune and dampen it as appropriate.

Create a deadened environment using a surround of acoustic screens.

Set up one drum at a time within this dead space and find the appropriate microphone and position.

Reassemble the kit using the close mic techniques found.

Listen for spill and reposition microphones accordingly, without compromising the sound.

Test overhead microphones and positions.

Test M/S microphone and positions.

Test ambient microphones and positions.

Experiment with processing.

It seemed to me that with two days meticulously analysing the drum kit in this way, it should become apparent exactly which microphones are more fit for task than others, and what techniques add to or detract from the overall sound.

Snare Drum

The snare drum was set within the screen and a selection of microphones aimed at it, in order to approximate recommended techniques suggested by various engineers over the years who may or may not have engaged in this kind of direct comparison. In total 8 mics were used, with diaphragms aligned as closely as possible, each pointing fairly flat against the drum head, around 1½ inches from it. This position seems logical, since aligning the diaphragm parallel to the drum head allows the greatest frequency energy of the drum as a resonating system to drive the diaphragm of the microphone, without passing obliquely across it, as is the case in a non-parallel placement. Indeed, adjustments to this end found that aiming the microphone at the point of stick contact, whilst possibly capturing fractionally more attack, does seem to incur a slight bass roll-off, which is a particular concern with condenser mics where this content is particularly important (since their sound is very different to the dry, bone-headed “thump” delivered by a dynamic).

During the test, each mic was analysed relative to the others for its particular qualities and ranked in order of “likeability”, based on its character and its frequency content. The results were as follows:

Microphone

Rank

Comments

Shure Beta 98

1

Bright, full bodied sound. Very directional but with an impressive amount of high end detail and a surprisingly thuddy bottom end. Everybody’s favourite by a long way. Its small size and gooseneck clip also makes it excellent for positioning.

Audio Technica ATM450

2

Another bright mic, although lacking a little of the detail and body of the Beta 98. Still a very good mic but may work better in combination with a dynamic mic to reproduce the depth of the drum (the Beyerdynamic M201 is an excellent dynamic microphone but unfortunately one was not available today).

AKG C414

3

A good, rich, unimposing tone, although with less “wow” factor than the aforementioned microphones. However, its physical size and cost render it probably inappropriate for all except the most affluent of engineers.

Josephson E22

4

The darling of Electrical Audio. A surprisingly dark sounding small diaphragm condenser microphone – its advantage is its rigidity and its directionality, however its top end response seemed to leave something to be desired.

Shure SM7

5

The brightest of the dynamic mics used, but still ranking below all condensers in terms of overall detail. Again, its physical size makes for problematic application in a real world scenario.

Shure SM57

6

Peculiar sounding dynamic microphone that has somehow found its way into the recording aether as a “workhorse” multi-application standard. In this test the SM57 ranks far below every other microphone so far discussed, with a very boxy, artificial sound that carries no discernible benefit other than the fact that you can happily throw it at a wall and find it still sounds the same.

Shure Beta 57

7

Even worse than the SM57, this microphone imparts a very noticeable high mid boost that forces its own character upon the source sound, producing a Frankenstein reproduction of a sound that could never exist in nature. A very poor mic.

Electrovoice RE20

8

This microphone ranks lowest only in as much as it was simply tried as a “bit of fun”. Dark, lacking in high end, physically large and intrusive, and expensive, this microphone is completely and utterly inappropriate for this application in every conceivable way.

With the most appropriate top-of-snare microphone now established, the next job was to determine the bottom mic. It therefore stood that in order to determine this, the designated top mic – of which we had only one – should remain on top whilst bottom microphones were tried in conjunction with it, in order that we could assess which complimented the top mic the best. We tested 7 microphones at the bottom of the snare, each time inverting the polarity (not phase – if you are of the school who think that phase and polarity are interchangeable terms then shame on you, and I will shortly be having words with your mother) so as to properly manage the conditions inherent in two microphones at close range pointing at each other. Taking a small moment to elaborate on this, because the bottom microphone receives the vibrations from the resonating drum head 180° out of phase with the top mic, either one of the corresponding channels should be polarity inversed, thereby adding further definition to the sound of the drum whilst avoiding low frequency cancellation, and also serving to cancel out any extraneous environment noise (like traffic or bass amps), since such noise arrives at the two microphones more or less simultaneously. This is known as a differential principle.

Here are the results of that test:

Microphone

Rank

Comments

AKG C414

1

Surprisingly complimentary sound with great separation and nice, rich body, rendering the overall drum sound extra punchy. A good range all round with a top end that is not too harsh. Very easy to place underneath the snare so physical limitations are no issue. Easily the best choice.

Audio Technica ATM450

2

Nice and bright (which is what you want in a bottom-of-snare mic) with a fast response, but lacking the depth of the C414. Still provides a good deal of snap.

Josephson E22

3

A slightly darker, slightly drier version of the ATM450. A good mic but not first choice in this application.

Shure SM7

4

Spits out a pleasant thump but lacks the top end detail of the aforementioned mics. Sounds as you might expect a dynamic mic under a snare to sound.

Sennheiser 441

5

Slightly richer than the SM7 but lacking in top end, rendering it fairly uncomplimentary to the top mic in this application.

Shure SM57

6

Characteristically dry, nondescript sound, not really useful for anything. Less thumping than you might expect and exhibiting extremely poor top end clarity.

Shure Beta 57

7

Entirely ridiculous. All artificial mid-range and nothing else. Sounds like a mechanical sneeze.

Although the microphone positioning in this case was based on identical principles as on the snare drum, the analytical process for the rack tom was slightly different, because essentially the mic that is fit for purpose on the top of the drum should also be fit for the bottom, given that the source is more or less the same:

Microphone

Rank

Comments

Josephson E22

1

This is where this mic comes into its own. Great attack and full body, nicely preserving the detail of the drum. The darker tone serves to enhance the tom sound very nicely without accentuating any irritating wolf tones.

AKG C414

2

Another nice, rich sound, with slightly less attack than the E22, but still pleasantly full-bodied. However, again it is physically intrusive and makes a great target for a drum stick.

Audio Technica ATM450

3

Slightly less body but good attack. Still a nice clean sound, but possibly not the first choice should an abundance of E22s be available. Having said that, there is absolutely not £500 worth of difference between this mic and the E22, and given its price, small profile and excellent position-ability, this is a great little microphone.

Sennheiser 441

4

A bone-head dynamic sound – nondescript and characterless, but “gets the job done”, if what you are looking for is the lower mid thump without concern for detail.

Shure SM7

5

More attack and less body than the 441. It’s passable, but it would be a strange engineer who considers this an appropriate microphone for any drum.

Shure Beta 98

6

All attack and no body. Strange, because it sounded so rich on the snare drum, but for a rack tom, this mic lacks the required depth to sufficiently reproduce the desired boom.

Shure SM57

7

Sounds like an SM57 – dull and lacking any redeeming features other than there tends to be lots of them about.

Shure Beta 57

8

Quickly becoming the stupidest sounding microphone. More imposing mid-range nothingness and entirely inappropriate for use on a drum kit, or, I suspect, anywhere else.

All in all, the E22 was clearly the most appropriate microphone for the rack tom, however, given their price at around £700, the likelihood of owning enough of them with which to entirely coat a drum kit is very small, unless you happen to be the man who conceived the design. We were lucky to have one to try out, but given the price difference of around £500, there is nothing whatsoever wrong with using the ATM450, which is lucky for us because we have an abundance of them. Where the E22 really scores points however is in its robust design and (as we were to discover later) its superior directionality, meaning that spill from the rest of the kit is less of an issue. However, the ATM450 has pretty good off-axis response, meaning that, even though it does pick up the rest of the kit more than the E22, it never actually sounds bad. Just be careful that your drummer doesn’t smack it with her stick.

It is probably unsurprising that the floor tom test harboured almost identical results to the rack tom test, save for some inconsequential ambiguity regarding whether the ATM450 was actually better than the C414 this time around. Either way, It’s pretty clear by now that if you are financially privileged enough to own enough E22s, this would almost certainly be the tom mic of choice, otherwise the ATM450 comes a close second. Either way I believe that you are going to end up with a close drum sound that boasts superior depth and clarity to the standard practice of using only dynamic microphones on the top heads.

When we came to examine the kick drum it was quickly established that the position of the mic had more bearing on the sound than the mic that was used. While every mic exhibited its own character, each of which could potentially be useful in different applications dependant on the style of music, the positioning harboured radically different results. Firstly, here is a brief analysis of the four mics tested at the hole of the kick drum:

Microphone

Rank

Comments

Electrovoice RE20

1

A great, full bodied microphone that really highlights the accompanying “boom” of the drum, as well as the attack sound. However a great deal of gain is required to produce suitable performance from the mic, and therefore decent preamps are advised that are suitable for the task.

AKG D112

2

Toppier than the RE20, it is actually a matter of preference which one you may decide to go for, with equally good attack and a mid-range that sounds… just… different to the RE20.

Audix D6

3

Apparently designed as a “metal” kick microphone, and you can definitely hear it, emphasising as it does the clicky attack that is a staple of irritating metal music everywhere.

Shure Beta 52

4

Not dissimilar to the D6, it only ranks lower because its physical shape renders it slightly harder to position.

The problem, however, was that, in the regular placement – just at the sound hole at the point where the greatest volume of air is being ejected from the drum – all of these microphones sounded peculiarly plasticky and weird. When listening to the drum in the room it sounded large and booming and just as you would hope a kick drum would sound, but under the microscope this was not the case. In order to remedy the situation, an AKG C414 was slowly moved around the outside of the repeatedly pounded kick drum while we determined to find the sweet spot at which the non-plasticky boom could be sufficiently captured. This transpired to be around 2 feet in front of the sound hole, where this microphone was subsequently positioned. This did bring to mind the idea that I now recalled from other engineers, that a condenser mic outside of the kick drum could sound quite nice, and here we seemed to have confirmed it. It had lots of attack and lots of boom, but the only component potentially missing was some extreme bottom end. Use of a Yamaha Sub Kick right against the resonant head seemed to provide the thud we were looking for, and when compared to the C414 with artificially boosted low end, it resulted in a much more natural sound.

An important component of a good kick drum sound however, and one that is almost always overlooked and then attempted to be fixed at the mix stage (usually by boosting somewhere around the 1-3 KHz mark) is the attack. It is this that is clearly audible in recordings, not the sub. The sub is something you feel, but the attack is something you hear. This is an important distinction to make, because I have seen many people (myself included, some years ago) trying in vain to boost the sub part of the kick drum, ever wondering why they are not able to allow it any definition through the sub frequencies of the rest of the mix. We all know the feeling of the bottom half of our mixes turning into a definition-less squelchy mulch of non-descript boom, without ever really being able to tell what is happening down there. And so the solution to this is to take care to faithfully capture the attack component of the kick drum by separately miking it. The bottom end of this signal can then be rolled off in order to achieve a piercing beater slap that accentuates the kick sound without ever having to boost silly frequencies in the mix. Obviously for this application a suitable microphone is any that has good high end definition but is also easy to place in such an awkward position. Three mics were tried in our test:

Microphone

Rank

Comments

Audio Technica ATM450

1

Great attack and full bodied, plus physically small. Combined with an Audix D-Clamp mounting clip, this is very easy to manoeuvre into a very close position. Just be sure it doesn’t get thudded by the beater. That wouldn’t be good.

Shure Beta 98

2

Excellent attack but lacking definition lower down in the spectrum. Also comes with a clip that looks like it is purpose built for this exact application, but after some frustrating fiddling you find that it really isn’t.

Sennheiser MD421

3

Boring dynamic nothing-special microphone that did nothing special in this application except got in the way.

Having determined the best microphones and relative positioning, the kit was assembled and all techniques applied with a view to checking how spill from the rest of the kit affected our choices. Happily, spill was hardly an issue, and in fact the benefit of dual miking drums was further accentuated in this regard, as having two microphones in different locations pointing at a single drum serves to enhance the character and definition of that drum whilst cancelling spill from elsewhere in the kit, or at least widening the relative signal-to-noise ratio. I suspect that this is therefore a critical consideration in using condenser microphones as close mics on a drum kit because, although sonically superior to dynamics, the advantage of dynamic mics is their off-axis rejection and poor high frequency response, meaning that crashing and clattering cymbals from elsewhere on the kit tend to interfere less with the dull character of the dynamic microphone. Therefore, as a rule of thumb, for excellent drum definition with condensers, use two of them.

The only instance in which spill was a noticeable concern was with the C414 in front of the kick drum, which by now was reproducing less of the boom exhibited by a single kick drum being repeatedly and consistently struck on its own, and more of the cymbal clatter from the rest of the kit. It had lost definition, and it was apparent that a closer technique was required in order to produce acceptable separation. It quickly became clear that a sound hole microphone could actually be used, so long as it was angled in such a way as to minimise the plastic tone of the beater slapping against the drum head. This was achieved by simply moving the mic off axis in relation to the beater. Still slightly plasticky, implying that there is some as yet undisclosed issue with the internal acoustics of that particular kick drum, but far less problematic than before, and with acceptable depth and precision. An RE20 was used in this application.

Overheads

Overheads were assessed in two ways; by exchanging microphones and by raising their position above the kit. The fundamental technique however always remained the same: both microphones positioned in a straight line either side of the kit, face-down directly above their respective cymbal clusters. Incidentally, it was a point of contention between Chris and I regarding which overhead microphone is considered “left” and which is “right”, with my firm belief that, as a drummer myself, it is appropriate for the stereo spectrum in the mix to be situated in a way that makes sense to the drummer – a right-handed drummer has his high-hat on the left-hand side. Anything else is disorientating and conducive to travel sickness. I fully accept, however, that not everyone shares this view, and some (most, probably) consider it appropriate to balance the mix from the audience’s perspective, rather than the musician’s. Personally my view is that the music was created entirely from the musicians’ perspective, from the composition to the lyrics to the performance, and therefore the recording and mix should reflect this and not be tailored to the perspective of an abstract person whose role is no more than an observer. I understand that a gig scenario has different connotations because a large audience is actually present and looking at the band, and therefore probably expects that their visual reference correlates with what they are hearing, but a recording is a musician’s opportunity to speak, unobserved, directly to an audience from their perspective. That’s why, to me, the high-hat side (for a right-handed drummer) is always OH L, and the floor tom side is always OH R. But whatever side of the fence you fall on, just make sure that you and any assistant engineers are all on the same page. It could lead to confusion.

In positioning the overheads, it was found that, for a hard hitting drummer such as myself at least, close miking the cymbals was inappropriate in that the attack part of the cymbal felt ear-splittingly harsh, whilst the sustain was largely unrecognised by the microphone. Not only this, but, as is often the case when overheads are too close to the cymbals, the movement of the cymbal creates an undesirable phasing effect. The remedy to this was to raise the microphones up to around four feet above the cymbals, being sure at all times that both microphones were equidistant from the snare, such that the risk of snare phasing could be minimised (an XLR cable makes a handy measuring device for this task). In this position, although allowing more room ambience into the microphone (which, in a room that sounds as functional as this, poses no problem), a good balance of the kit could be achieved with great stereo imaging. There was talk of elaborating on the positioning by trying a coincident pair or some such oft-referred to but curiously never observed technique, but given that the stereo imaging was doing everything we needed it to do here, it seemed inappropriate to change it.

The mics tested were as follows:

Microphone

Rank

Comments

Coles 4038s

1

Deceivingly dark sounding microphones, such that on first glance they appear entirely inappropriate until you realise that they are perfectly balancing the kit and smoothing out all the aggressive high end usually present in overhead microphones. Very dry and functional sound without excellent body.

Neumann U87s

2

A bright, clattering sound, rich in upper mid detail but inappropriate for this application without significant EQ treatment.

Neumann KM187s

3

Very bright with excellent high end detail and transient response, but as such renders clattering cymbals ear splitting.

An essential component of the “Albini” drum technique is the placement of a cardioid/figure-8 arrangement in front of the kit, with the side mic aligned to face the walls perpendicular to the direction of the kit. The principle of M/S (mid/side) recording is that, whilst a directional microphone captures the sound source, a figure-8 mic captures a stereo image by means of polarity-inverting a duplication of its signal, the two of which are then hard panned left and right. And so it was in this test that, after experimenting with the positioning of the mic in front of the kit, moving it increasingly further back until the most desirable spot was found (around four feet from the kit and 2½ feet from the floor), the resulting trickery harboured subtle ambience which lent itself nicely to the width of the kit. It was however important the right microphones are used, as the initial choice of Audio Technica AT4050 – although a great microphone for many instances – actually sounded too bright and revealing here. Cymbal clatter is a large problem in ambient drum miking, and so use of ribbon mics – in this case the Royer 121 – provided just the right amount of high end roll off that served the sound well, although ultimately, at mix stage, the high end had to be further attenuated to minimise the clanging clutter as much as possible.

Ambient Mics

With the essential sound of the kit now nicely built up, and the drums thumping through the speakers in a way that has not been observed by any of us in any studio this side of Chicago, ambient miking became merely a matter of taste, the essential function of which was to add a little extra something into an already more than serviceable sound. In this regard, we spent little time experimenting with different microphones and placements, but instead went with a tried and trusted technique of using two fairly transparent sounding omni mics (in this case some custom built Panasonic WM-61s – two surprisingly high quality electret capsule mics that I built some time ago) placed ¾ of the way up the wall at the back of the room, thus utilising the principle of the boundary effect. Boundary microphones exhibit more accurate definition due to their being uninhibited by the comb filtering effects caused by wave cancellation from nearby boundaries – placing the mics on the boundary itself reduces such an undesirable effect.

Following this, a rusty, vintage tube microphone was placed in the corridor outside the live room in order to capture what I have come to regard as “a bit of fun” – special effect ambience that you can take or leave, depending on taste. Just compress the crap out of it and – presto – instant drum fun.

And so essentially, that was it. All this experimenting resulted in a drum sound that, when played back, just about knocked your head through the rear wall, which is precisely the effect we were striving for. If you would like to hear for yourself, you can find the resulting files here.

For the time being, anyway, I’m off to give my fingers a rest from about three hours’ constant tip-tapping. In another post I will elaborate on the mixing process following these techniques, as one or two tricks should be employed in order to suitably manage such a large number of independent signals. However, for the time being, this appears to be the secret of the Albini method.

My name is James Gasson. I am a musician, sound engineer, artist and chief operator of Third Circle Recordings. I journey through life trying to work out what exactly is going on whilst doing my best to avoid tripping over. Some days are more successful than others.

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